Telecommunications facilities equipped with a so-called “hands free” function are often sensitive to acoustic echo. For example, in the case of a telephone facility, acoustic echo occurs when the signal emitted by the loudspeaker, which corresponds to a signal uttered by a remote talker, is picked up by the microphone. This is then manifested, for this remote user, by the reproduction, in the earpiece, of the signal with an offset with respect to the instant of emission.
Likewise, the presence of a two wire/four wire line transformer in the remote telecommunications facilities, the function of which is to process the signals received so as to render them compatible with the telephone line, is apt to bring about an electrical echo by returning over the line a part of the signal received.
The electrical echo or the acoustic echo is apt to disturb communications when the delay in the chain becomes significant.
Reduction, or even cancellation of the echo, is achieved by equipping the facilities with an echo canceller device AEC or EEC. This type of device generally incorporates an adaptive filtering algorithm whose coefficients are calculated in such a way as to minimize the error between the echo and an estimate of the echo. The coefficients are calculated in a recurrent manner on the basis of previously calculated coefficients, of the error between the estimate of the echo and the echo, of the reference signal and of an adaptation stepsize μ, on the basis of the following relation:H(n+1)=H(n)+μ(n)*error(n)*f(X(n))in which:                H(n+1) and H(n) represent the coefficients of the adaptive filter at the instant n+1 and at the instant n, respectively;        f(X(n)) is a function of the vector of the last L samples of the signal emitted X(n), varying according to the families of algorithms considered;        μ(n) represents the adaptation stepsize;        error(n) represents the distance between the real echo and the estimated echo.        
Represented in FIG. 1 is the general architecture of an echo cancellation module, according to the state of the art. Such a module incorporates an adaptive filter 10 which estimates, on the basis of the signal X(n) emitted by a facility, the electrical or acoustic echo E and which uses a subtractor 13 to subtract the estimate of the echo from a signal received.
As is known, such a filter constitutes a highly recursive system, requiring appropriate supervision to guarantee its stability and to obtain convergence. In fact, the adaptation of the coefficients according to the equation mentioned hereinabove should be done only in the presence of echo alone so as to correctly estimate the real echo, that is to say in the absence of simultaneous local speech. If this condition is not fulfilled, the adaptation will not be done correctly and, in the limit, the filter may become unstable and transform itself into a noise generator. This sensitivity to instability is aggravated by the speed with which the algorithm converges: a fast algorithm will converge speedily towards an optimal filter if the signal originating from a microphone contains echo only, but will diverge equally speedily if another signal, for example speech, is superimposed on the echo signal.
To alleviate this drawback, various techniques may be used, in such a way as to curb the adaptation of the filter, that is to say to permit the updating of the coefficients only in a situation of echo alone.
A first technique consists in using detectors of vocal activity to determine the state of the system. The adaptation strategy is applied as a function of the state of the system, in such a way in particular as to instigate the adaptation of the coefficients of the filter only in the presence of echo alone.
Other techniques consist in transforming the adaptation stepsize, which is generally fixed, into a parameter tending to zero in the presence of speech, in such a way as to halt the adaptation of the coefficients of the filter. To this end, reference may be made to French patent application 96 05 312 in the name of the applicant.
The document JP-A-2003 324370 describes a comparable system based on the calculation of the energy ratio between the signal of the loudspeaker and the signal of the microphone.
Such a system makes it possible to decrease the value of the adaptation stepsize for the coefficients of the filter, and hence the convergence of the filter. However, in no case does it make it possible to detect a divergence thereof.
Finally, other techniques implement a comparison of the error after convergence with the level of the background noise, so as to halt the adaptation and, thereby, to prevent the divergence of the filter. To this end, reference may be made to U.S. Pat. No. 5,477,535, in which the adaptation of the filter is disabled when the error between the estimate of the echo and the echo reaches a threshold value, so as to avoid any divergence of the filter.
According to these various techniques, the main aim is to prevent any divergence of the filter. No solution is advocated when divergence of the filter occurs despite everything, and such divergence may indeed occur in particular situations of prolonged double speech, that is to say in the presence of local and remote speech, or of very abrupt variations of acoustics in front of the terminal, for example when a sheet of paper is placed over the mic of the terminal.